[asterisk-bugs] [JIRA] (ASTERISK-29618) ConfBridge errors on creation conference room
Alexander (JIRA)
noreply at issues.asterisk.org
Mon Aug 30 10:02:33 CDT 2021
[ https://issues.asterisk.org/jira/browse/ASTERISK-29618?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]
Alexander updated ASTERISK-29618:
---------------------------------
Description:
After Asterisk upgrade from 16.9.0 to 16.20.0, we began to experience problems with the application when creating ConfBridge conference rooms for 2+ participants. Video streams for some participants do not appear.
{code:title=logs|borderStyle=solid}
[Aug 25 11:11:53] ERROR[106619][C-00000001]: bridge_softmix.c:669 sfu_topologies_on_join: CBAnn/conference_room_37197-00000000;2: Couldn't request topology change
[Aug 25 11:12:08] ERROR[106435]: res_pjsip_session.c:2158 resolve_refresh_media_states: State not consistent
[Aug 25 11:12:08] WARNING[106435]: res_pjsip_session.c:2299 sip_session_refresh: PJSIP/webrtc-00000000: Unable to merge media states
{code}
{code:title=extensions.conf|borderStyle=solid}
[conference_room]
exten => create,1,NoOp('Create conference room')
same => n,GoSub(add_member,1)
exten => add_member,1,NoOp('Add member to conference room')
same => n,GoSub(configure_client,1)
same => n,GoSub(configure_bridge,1)
same => n,GoTo(join,1)
exten => configure_bridge,1,NoOp('Configure bridge profile')
same => n,Set(CONFBRIDGE(bridge,template)=conference_members_bridge)
same => n,Return()
exten => configure_client,1,NoOp('Configure client params')
same => n,Set(user_profile=client_profile)
same => n,Set(user_menu=client_menu)
same => n,Return()
exten => join,1,NoOp(Join to ConfBridge: conf_name=${conf_name}, user_profile=${user_profile}, user_menu=${user_menu})
same => n,ConfBridge(${conf_name},,${user_profile},${user_menu})
same => n,Hangup()
{code}
{code:title=confbridge.conf|borderStyle=solid}
[client_profile]
type=user
end_marked=yes
jitterbuffer=no ; Jitterbuffer should be disabled when video is used.
quiet=yes
startmuted=no
dsp_drop_silence=no
talk_detection_events=no
[conference_members_bridge]
type=bridge
video_mode=sfu
max_members=15
record_file_timestamp=no
mixing_interval=10
video_update_discard=1000
remb_send_interval=1000
remb_behavior=average_all
{code}
Steps:
{code:title=ARI|borderStyle=solid}
1. Add first user to the conference:
POST http://asterisk.node:8088/ari/channels
{"endpoint": "PJSIP/81014504875 at webrtc", "callerId": "<8007709999>", "context": "conference_room", "extension": "create", "variables": {"conf_name": "conference_room_37368", "formats": "opus,vp8" ... }
2. Add other participants. For example:
POST http://asterisk.node:8088/ari/channels
{"endpoint": "PJSIP/81014504876 at webrtc", "callerId": "<8007709999>", "context": "conference_room", "extension": "add_member", "variables": {"conf_name": "conference_room_37368", "originator": "37368" ...}
Where "originator": "37368" - first user's channelId
{code}
was:
After Asterisk upgrade from 16.9.0 to 16.20.0, we began to experience problems with the application when creating ConfBridge conference rooms for 2+ participants. Video streams for some participants do not appear.
{code:title=logs|borderStyle=solid}
[Aug 25 11:11:53] ERROR[106619][C-00000001]: bridge_softmix.c:669 sfu_topologies_on_join: CBAnn/conference_room_37197-00000000;2: Couldn't request topology change
[Aug 25 11:12:08] ERROR[106435]: res_pjsip_session.c:2158 resolve_refresh_media_states: State not consistent
[Aug 25 11:12:08] WARNING[106435]: res_pjsip_session.c:2299 sip_session_refresh: PJSIP/webrtc-00000000: Unable to merge media states
{code}
> ConfBridge errors on creation conference room
> ---------------------------------------------
>
> Key: ASTERISK-29618
> URL: https://issues.asterisk.org/jira/browse/ASTERISK-29618
> Project: Asterisk
> Issue Type: Bug
> Security Level: None
> Components: Applications/app_confbridge, Resources/res_pjsip
> Affects Versions: 16.20.0
> Environment: CentOS Linux 7.4.170, kernel 3.10.0-1127.13.1.el7.x86_64, libsrtp 1.5.4
> Reporter: Alexander
> Assignee: Unassigned
> Labels: webrtc
> Attachments: aster_cli_16.20.0.log, aster_cli_16.9.0.log, debug_log_29618_16.20.0_bad_call.log, debug_log_29618_16.9.0_good_call.log
>
>
> After Asterisk upgrade from 16.9.0 to 16.20.0, we began to experience problems with the application when creating ConfBridge conference rooms for 2+ participants. Video streams for some participants do not appear.
> {code:title=logs|borderStyle=solid}
> [Aug 25 11:11:53] ERROR[106619][C-00000001]: bridge_softmix.c:669 sfu_topologies_on_join: CBAnn/conference_room_37197-00000000;2: Couldn't request topology change
> [Aug 25 11:12:08] ERROR[106435]: res_pjsip_session.c:2158 resolve_refresh_media_states: State not consistent
> [Aug 25 11:12:08] WARNING[106435]: res_pjsip_session.c:2299 sip_session_refresh: PJSIP/webrtc-00000000: Unable to merge media states
> {code}
> {code:title=extensions.conf|borderStyle=solid}
> [conference_room]
> exten => create,1,NoOp('Create conference room')
> same => n,GoSub(add_member,1)
> exten => add_member,1,NoOp('Add member to conference room')
> same => n,GoSub(configure_client,1)
> same => n,GoSub(configure_bridge,1)
> same => n,GoTo(join,1)
> exten => configure_bridge,1,NoOp('Configure bridge profile')
> same => n,Set(CONFBRIDGE(bridge,template)=conference_members_bridge)
> same => n,Return()
> exten => configure_client,1,NoOp('Configure client params')
> same => n,Set(user_profile=client_profile)
> same => n,Set(user_menu=client_menu)
> same => n,Return()
> exten => join,1,NoOp(Join to ConfBridge: conf_name=${conf_name}, user_profile=${user_profile}, user_menu=${user_menu})
> same => n,ConfBridge(${conf_name},,${user_profile},${user_menu})
> same => n,Hangup()
> {code}
> {code:title=confbridge.conf|borderStyle=solid}
> [client_profile]
> type=user
> end_marked=yes
> jitterbuffer=no ; Jitterbuffer should be disabled when video is used.
> quiet=yes
> startmuted=no
> dsp_drop_silence=no
> talk_detection_events=no
> [conference_members_bridge]
> type=bridge
> video_mode=sfu
> max_members=15
> record_file_timestamp=no
> mixing_interval=10
> video_update_discard=1000
> remb_send_interval=1000
> remb_behavior=average_all
> {code}
> Steps:
> {code:title=ARI|borderStyle=solid}
> 1. Add first user to the conference:
> POST http://asterisk.node:8088/ari/channels
> {"endpoint": "PJSIP/81014504875 at webrtc", "callerId": "<8007709999>", "context": "conference_room", "extension": "create", "variables": {"conf_name": "conference_room_37368", "formats": "opus,vp8" ... }
> 2. Add other participants. For example:
> POST http://asterisk.node:8088/ari/channels
> {"endpoint": "PJSIP/81014504876 at webrtc", "callerId": "<8007709999>", "context": "conference_room", "extension": "add_member", "variables": {"conf_name": "conference_room_37368", "originator": "37368" ...}
> Where "originator": "37368" - first user's channelId
> {code}
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